Friday, December 29, 2006

What's going on with Microsoft / Nortel / Mitel

Allright, things are getting a little crazy in VoIP world.

Microsoft's new Exchange 2007 server can be a Unified Messaging Server / Media Server, Live Communications Server (soon to be Office Communications Server) is an Instant Messaging / Presence and SIP proxy and both Nortel and Mitel are claiming to be the 'chosen' ones working with Microsoft.

Nortel outlines roadmap for Microsoft partnership

Microsoft Technology Centers Showcase Unified Communications with Mitel Solution

I think both vendors see the writing on the wall... With 60% of the enterprise mail server market, Exchange is a force to be reconed with. Let's face it, a large portion of that will upgrade to Exchange 2007 at some point. Why, if I have Exchange and have already purchased Exchange CALs and it will work as my VM server, should I go and spend big $ on a different UM solution and fight the integration battle.

MS Outlook owns an even higher percentage as the e-mail client of choice.

Microsoft is trying to position Office Communications Server (LCS follow-on) and the Office Communicator in the same mold. Office Communicator may become the softphone of choice as Outlook is the e-mail client of choice.

It is a pretty easy bet that MS will continue to own the desktop software market. The traditional phone vendor world has also done a terrible job with their desktop integrations. I don't think there will be a question as to whether MS can rule the desktop softphone market and thus drive their server software LCS sales.

I think the question will really be, how long will Microsoft need to keep those relationships around... if they are really even "real" relationships to begin with or just old school vendors trying to hang on to those licensing dollars.

Tuesday, October 31, 2006

Pingtel Announces installing sipXchange ECS

Pingtel has annouced that will be running their communications through Pingtel's Open Source sipXchange Enterprise Communications System (ECS).

The deployed system is a High Availability implementation utilizing 2 registration / authentication proxy servers with a separate media server for voicemail and auto attendant.

Monday, October 23, 2006

Windows Mobile 5.0® SIP-Based Dual-Mode Phone: hipi-2200™

Paragon Wireless introduces the world’s first SIP-based Windows Mobile 5.0® dual-mode handset—the hipi-2200. The hipi-2200 is the next generation in integrated dual mode handsets. Utilizing the proven VoWLAN / GSM dual-mode voice technology, the hipi-2200 adds the Windows Mobile 5.0® operating system to further expand the usability and functionality of this extremely versatile handset. With the built in 2.0M pixel camera, the MP3/MP4 player and the camcorder/recorder on a Windows Mobile® platform, users can access the Internet, listen to music and take and review pictures all from one easily carried handset, that also serves as their mobile phone. The 1100mAh Li-ion battery offers users up to 4 hours of talk time, or 100 hours of standby time, making the hipi-2200 the only handset they need to carry.

Here's a link:

New Polycom 650 phones

Saw and played with these at VON. Sound quality is awesome!

Here's some of the marketing stuff...

Revolutionary Voice Quality The SoundPoint IP 650 is the first IP phone to use Polycom’s revolutionary HD Voice technology that delivers voice communications of life-like richness and clarity.

Advanced Features and Applications The phone’s SIP 2.0 software fully supports Microsoft Live Communications Server 2005 for telephony and presence, and integrates with Microsoft Office Communicator instant messenger client. The SoundPoint IP 650 features a USB port for future applications.

Enhanced Call Handling Capabilities The SoundPoint IP 650 accommodates 6 lines in a standalone mode and up to 12 lines when equipped with SoundPoint IP Expansion Modules, as an attendant console. The phone supports shared call / bridged line appearances as well as busy lamp field (BLF) functionality that enables phone attendants to monitor and manage calls more efficiently.

Expandability When equipped with up to three SoundPoint IP Expansion Modules, the SoundPoint IP 650 delivers the advanced call handling capabilities and enhanced user interface of a high-performance attendant console. (note, haven't seen the updated expansion modules yet... I assume they will be backlit as the new phone is)

Intuitive User Interface The SoundPoint IP 650 delivers all of its capabilities through an intuitive user interface, featuring a high-quality backlit 320x160 graphical grayscale LCD display, an easy-to-navigate menu, and a combination of dedicated keys and context-sensitive soft keys for one-button access to essential telephony features.

Fonality Acquires trixbox

LOS ANGELES, Calif. - October 04, 2006 - Fonality, the leader in IP telephony systems for small and medium businesses and the world's largest commercial Asterisk based deployment, today announced it has acquired trixbox, the world's largest Asterisk based community. trixbox founder Andrew Gillis will join Fonality and continue to lead the trixbox community. Fonality will commit engineering resources and broad financial support to continue fostering innovation in the trixbox open source community.

See full press release here:

Thursday, August 17, 2006

Creator of Ethereal(R) Joines the WinPcap team; Wireshark is Born

Davis, CA (PRWEB) June 8, 2006 -- We are proud to announce that Gerald Combs, creator of Ethereal®, has joined CACE Technologies ( He will be working with Loris Degioanni and Gianluca Varenni, the creators of the WinPcap packet capture library (, forming a world-class team of network analysis experts. As his first venture in this new alliance, Gerald has created the Wireshark network protocol analyzer, a successor to Ethereal®.

Wireshark's home is Enhanced and improved, Wireshark is the ultimate tool of choice for network troubleshooting, protocol development, and education worldwide. The unique partnership of Wireshark and WinPcap brings a new synergy, power, and benefits to the open-source community and industry. The upcoming version of Wireshark will be 0.99.1. A pre-release version is available right now at

"I am indebted to core development team of Ethereal® for joining me to work on Wireshark. With their help and contributions from the user community, we're set to continue our success in building the world's leading open-source network protocol analyzer. We have lots of new and exciting things planned for Wireshark! I'm also really excited about joining CACE. Loris and Gianluca are well respected in the community, and it will be great to work with them. As an added bonus, Davis is a great place for my wife and me to raise our daughter," said Gerald.

"We’re thrilled to welcome Gerald to CACE Technologies and expect to do great things together. The sky’s the limit," said Loris.

ABOUT CACE Technologies - CACE Technologies,, is an innovative and dynamic company specialized in low-level networking solutions. We are experts in Windows and Linux device driver and network monitoring tools development.

CONTACT: John Bruno, CACE Technologies, john.bruno @

Tuesday, July 11, 2006

Nerd Vittles Newbie's Guide to TrixBox 1.1 and freePBX

Today we'll show you how to install the latest and greatest TrixBox 1.1 with freePBX 2.1.1 in just over an hour. As with the earlier release of TrixBox, these new Asterisk products are designed to support the casual home or home office user's PBX needs as well as gigantic call centers processing millions of calls a month.

Everything is free except the hardware on which to run your new system. That can be almost any old Pentium PC or a multi-processor RAID box with mainframe horsepower. We also want to get TrixBox properly configured to support our next free application: TrixBox MailCall.
It'll let you retrieve and play back your email messages using any touchtone telephone and your TrixBox 1.1 system. And, yes, you'll need TrixBox 1.1 to make everything work.

To be fully nerded:

Monday, July 03, 2006

Voip Deployment - The Virtual Network

Virtual LANs (VLANs) have nothing to do with quality of service (QOS) in a VoIP / IP Telephony deployment. There, I said it.

So why would you bother use them? For the quality of the deployment (QOD? :-). By logically segmenting the voice and data worlds disruptions in either world will not affect each other (hopefully). You can firewall or use access lists between VLANs to help secure your VoIP deployment. Also, if you want DHCP / DNS to work differently for the phone system it doesn’t affect the data network.

I’m not going to get into the intricacies of VLANs here. Here’s the wikipedia entry ( which will help a bit. Get a couple of switches that support VLANs and play with them a bit… learn how to trunk VLANs between switches and how to statically map ports into VLANs. Learn them, live them, love them… you’ll use them all of the time.

On most VoIP deployments I’ll use 3 VLANs at a minimum. Data, Phone and Management. Data is the default VLAN and used for all PC’s, servers & printers. Phone is for all phones, gateways & PBX equipment. Management is for switch / router management IP’s.

Here’s what the VLAN diagram would look like:

Statically map the PBX and gateway ports into the Phone VLAN. Setup all of the ports that will connect to PC’s and phones with the Data VLAN as the default VLAN (untagged) and the Phone VLAN as a tagged VLAN. That means that devices ‘tagging’ their traffic for the Phone VLAN will be placed in that VLAN and devices that don’t know how to tag their traffic will be in the Data VLAN.

Manually set your IP phones to be in the Phone VLAN. This means the phone will tag it’s traffic for that VLAN and pick up DHCP from that VLAN. It is possible for some phones to pick this up off of the initial DHCP reply to the phone and then switch to the Phone VLAN and get another DHCP address from that VLAN. However, now you are depending on the DHCP server on the Data VLAN to be working.

Saturday, July 01, 2006

Voip Deployment - The Physical Network

So, you want to install a VoIP / IP Telephony system. How do you prepare your LAN / WAN for this new application where timing is critical? How well your VoIP system runs will depend largely on how good your network infrastructure is.

Cabling - Make sure your network cabling is up to standards. Properly run Cat 5/5e/6 cabling with patch panels in the closets & wall jacks on the wall. Use manufactured patch cords. And if you really want to be sure it is right, have it certified by a cabling contractor.

Network Switching - This is one of the harder pieces and critical to the success of your deployment. I won't pretend to be able to describe how to properly design a network in a few paragraphs because every organization is different. Here are some basics though.
  • Think about the core of your network. If everything ties back to a single location things are pretty simple. Where are your servers, your wiring closets, your wide area connectivity & your outside world connectivity? Start at the core and work out from there with a simple star topology (don't try to connect closet to closet to closet, instead connect each closet to the core directly).
  • What other critical line of business applications do you run that may need to be considered in the design.
  • What do you have for existing equipment that you might be able to utilize?
  • Figure out which manufacturer's gear you want to utilize and learn the different models and their options. This is important! Know your product!
  • How are you going to power your hard phones? PoE switches, mid-spans (power injectors) or power bricks at each desk?
  • How about power protection? Most UPS manufacturers have calculators on their web sites that allow you to estimate consumption and run times of their gear.

Here's an example of a physical network design with a core server room (the core closet area is sometimes referred to as the MDF and the remote closets as IDFs).

Above we see the two closets linked back to the core at 1 Gbps, one with fiber because of distance from the core, and the second with Cat 5e / 6 copper. The servers all connect in to the core network switch at 1 Gbps and the firewall and router connect in to the core. Simple, clean efficient.

I like the HP switching gear (it's about 60% of the cost of an equivalent Cisco design, lifetime replacement, support and software updates). They have a nice broad range of products, their QOS seems pretty good and they are easy to configure. Don't' get me wrong, I like the Cisco stuff too and design plenty of networks with it. I just think I get more bang for the buck with HP. If I need to cheap things out, Linksys has some inexpensive managed PoE switches and so does Dell, Netgear and DLink.

Try to stick with managed switches so that you can create VLANs. Most modern managed switches will support Quality of Service (QOS). Some of the really cheap PoE switches are unmanaged (Netgear, DLink have some models like that).

Adtran makes some nice little stackable PoE switches (1224 series) and they even have one with an integrated router module (1224 r). This makes for a nice all in one device at remote WAN connected locations.

Don't have the coin for all new fancy gear? First off, prepare yourself for small voice quality issues. If you can live with that (hey we all put up with cell phones right?), take the above network design principles into account. Good cabling, star topology, avoid linking switch to switch to switch. Oh, and forget about trying to use any old hubs you have... switches only.

Remember, cabling and switching are the foundation of your VoIP deployment. The rest of the house is only as good as the foundation. Take the time to get it right and people won't be reaching for your throat. Next article I'll get into VLAN design and maybe QOS... we'll see how long it runs...

Sunday, June 25, 2006


Snap is a cutting edge dialer and call pop up application for Asterisk and other IP PBX's that is flexible and powerful.

Snap works by sending the phone number you wish to dial to your PBX and initiating a call back to your phone. Once your phone rings you pick it up and it will be connected to the number you dialed.

For Outlook users, the call pop-ups are tightly integrated.The flexibility comes from it's Multi-Connection technology. If you travel between work and home, or would like to have seperate settings for different situations then this feature will be extremely useful for you. You simply dial via a different connection by using the "arrow" to the right of the Dial button to access these connections.

The latest version of the product (0.7) adds TAPI (in the Pro version $29.95) and eyeBeam 1.5 integration.

Link to their web site:

Wednesday, June 14, 2006

Paragon GSM / WiFi phone launched

Paragon Wireless launched a new GSM / WiFi called 'Hipi'. The new phone is a dual-mode phone that supports GSM (900, 1800 & 1900 MHz), GPRS and WiFi with SIP.

Paragon reports that the phone has been tested in the United States with some 30 different SIP infrastructures and providers.

The phone also sports PDA functionality with e-mail, calendar, a web browser and a camera (1.3 M pixel).

The software in the phone is based on Linux. At its core, the phone utilizes an Intel PXA271 processor.

Battery life was worked on extensively with the phone. Apparently 70 to 100 hours of standby time can be had, even with both radios turned on.

Here are the impressive published specs...

Paragon has recently opened an office in Dallas with two employees and has approximately 40 employees in China.

Saturday, June 10, 2006

Mitel launches two new phones

After getting beat up a bit in the market for not having any middle of the road 'DESI-less' IP phones, Mitel has introduced their new 5330 and 5340 IP phones. The new phones are attractively designed and will definately help Mitel stand out in the market.

Mitel phones have offered excellent sound quality in their previous models. Looking to raise the bar in audio quality, Mitel has added wide band audio support to both phones. The key is in the new handsets which can operate at 7 Khz.

As typical with all Mitel phones the new phones offer a high quality and solid feel.

Mitel 5330 - No backlight

Mitel 5340 - backlight

Both phones offer the following features:
  • Large graphics display (160 x 320)
  • 24 on 5330 and 48 on 5340 Programmable, multi-function, self-labeling keys, provided in three pages of 16 keys each (for one-touch access to speed calls, line appearances, features)
  • Wideband Audio Support – ships with a wideband handset (7kz) standard
  • Peripherals and modules support: Line Interface Module, IP Conference Unit, Wireless LAN Stand, Gigabit Ethernet Stand
  • 13 fixed function keys: Hold, Settings, Message, Speaker, Mute, Transfer / Conference, Redial, Cancel, Volume/Ringing/Contrast Up & Down, Home Page, Previous Page, Next Page
  • Six (5340) or Three (5330) context-sensitive softkeys for intuitive feature access
  • HTML Desktop Toolkit included for Applications development
  • PC Companion Application for easy user programming and key labeling
  • Dual mode phone: support for SIP and MiNET protocols
  • Handsfree speakerphone operation (full duplex)
  • Dual port IP phone (10/100 Mb integrated Ethernet switch)
  • Language Support: English, French, German, Italian, Portuguese, Spanish, Dutch
  • 802.3af power compliant (IEEE Standard)
  • Supports IEEE 802.1p/q for Voice Quality of Service
  • Designed for power conservation: reduces power consumption for overall energy savings

Thursday, June 08, 2006

Polycom Introduces new SoundPoint IP 430

Polycom full-duplex speakerphone and a robust feature set in a two-line phone

In what no doubt is the initial release in a new series of phones, Polycom has introduced the new SoundPoint IP 430. The 430 would be considered a replacement for the SoundPoint IP 300/301.

One of the pieces always missing in the 30X line (and the 50X) was true Power Over Ethernet (802.3af). A special cable had to be purchased with the phone in order to support PoE. This finally has been addressed in the 430.

The IP 430 is an enterprise grade piece of equipment designed for a typical cubicle worker. It has the solid feel that businesses demand and we have come to expect from Polycom. A robust set of features is available on the phone including:
  • Full duplex speakerphone featuring Polycom Acoustic Clarity Technology
  • Two 10/100 Ethernet Ports
  • 2 Lines
  • 3 Way Local Conference Calls
  • Amplified headset RJ-9 jack
  • SIP Protocol
  • Call park
  • Pick-up
  • Hold and transfer
  • Shared call / bridged line appearance
  • Multiple call appearances
  • Presence
  • Instant Messaging
  • Integration with Microsoft Live Communications Server 2005
  • New secure provisioning capabilities
With this refresh of their product line Polycom will solidify their position as one of the top IP phone manufacturers.

Read more here:,1443,pw-34-182-15672,00.html

Mitel's New IP Phones Based on TI's VoIP Technology

IP-based technology provider Mitel announced on Wednesday that its new suite of IP phones is based on Texas Instruments (TI)'s VoIP technology. The companies' partnership allowed for the integration of TI's TNETV1050 IP phone system with Mitel's (News - Alert) new 5300 series IP phones to deliver real-time access to applications and services, including Web-bed browsing, contact lists, call history logs, amongst others.

Read More: Mitel's New IP Phones Based on TI's VoIP Technology

Tuesday, June 06, 2006

Hitachi WiFi IP Phones - WIP 5000

Hitachi WiFi IP Phones WIP 5000

I've had one of these for a while and tried it successfully on sipX and Asterisk systems.

The Hitachi WIP 5000 WiFi phone is a nice quality set. It feels solid to the hand and not just like a cheap piece of plastic.

Wireless network setup is not for a noob but not overly difficult. Only open and WEP 64/128 are supported for modes. It would be nice if WPA was also available... most people get turned off when only WEP is supported.

The screen is a monochrome LCD with a backlite. Very easy to read and a reasonable size.

There is a cool optional USB cable available for charging the unit while on the road. There are also extra batteries & carrying case to go with the phone. Batteries are easy to change but a single battery should last most users a full day.

There's also headset jack on the left side of the unit allowing for hands-free operation.

As with any WiFi phone you'll want to watch the number of sets per access point but you should be pretty safe with 5 to 6. If your users aren't that active you can probably get away with more.

Monday, June 05, 2006

Counterpath eyeBeam 1.5.5

Counterpath has tweaked their successful eyeBeam 1.1 with a new version 1.5.

Version 1.1 of eyeBeam was a pretty solid product with good support for USB headsets... especially for the uber cool Plantronics CS-50 USB (can answer and hangup calls right from the wireless headset).

There were some minor annoyances however like when you double clicked on one of your contacts it would open up an instant message box instead of dialing the user. That is now a user selectable option in 1.5.5.

Speaking of contacts, the Calls and Contacts "wing" is now cleaned up a bit (call logs are on a separate tab rather than clogging up the Contacts page). Contacts can also now be importand from Outlook .pst files in addition to the .csv and vcard files.

There was also a bug in 1.1 now allowing for Consultative / Warm / Attended (I've seen it called all three) Transfer. This problem is now fixed in version 1.5.5.

The entire Options menu is now cleaned up. Device support seems simpler as does maintaining status messages for presence. Ringtones are now possible (.wav files only).

Another nice new feature is the ability to do Audio, Video & Signaling QOS usinging DSCP/TOS values. We usually setup HP gear in for QOS COS 6 (DSCP 48 is typically where COS6 maps in to by default) and Cisco gear for QOS COS 5 (DSCP 40, that's what Auto QOS does for Cisco phones so I usually just play along...).

One annoying change is that they swapped the buttons for pickup and handup. The green button is now on the left and the red 'hang-up' is now on the right... grrr....

All in all, I'd sait is's a worthwhile upgrade. If you're having issues with things like the Transfer function they'll probably even give you the upgrade for free.... nice folks that they are....

Sunday, June 04, 2006

Fonality Hud Lite on Trixbox

Finally got Hud Lite working this morning. It should be considered a companion for you desk phone.

It has its own web site at:

The trick to making it work is that your devices should all be named SIPx where 'x' is the extension. Also, use your deskphone's extension and SIP password.

The default HUD Lite server password is 'password'. To change it see this post:

HUD Pro is only available if you have a Fonality PBX. Apparently it only works with Asterisk 1.09 at the moment. It would be nice if they allowed you to purchase just HUD Pro but I guess they are counting it as one of their PBX's competitive advantages. It is a nice looking product and seems to work very well.

Saturday, June 03, 2006

trixbox initial impressions

Downloaded trixbox 1.0 on 6/1 and finally had time to install and test it this morning.

The install was very simple on to a system I've had running A@H 1.28 and sipX.

Once the CD installation was complete, logged in as root and ran netconfig to setup an IP address on the system.

From there I was able to hit the box from a web browser on another machine and work with freePBX as in version 1.28 of A@H. Love the new updating options with trixbox and freePBX. Hopefully it is the end of full re-installs. This was one of the big issues I had with the product before. That and the hangup with the Asterisk At Home name.

I like the looks of the new Hud Lite application on the pbx. Still fighting with trying to get it to function as a softphone (not sure if it even can yet). One thing I don't like is that the extensions you want to see in hud lite must be entered in separately from the extensions when setup in freePBX.

I'm also fighting with some call quality issues with Counterpath's eyebeam 1.1 with Asterisk. I had this issue with A@H 1.28 and they continue. The hard phones seem to work fine and eyebeam works fine with the sipX / Pingtel pbx I'm working on.

Thursday, June 01, 2006

New Adtran 7000

Adtran rep came into the office to give us a demo on the new 7000 series IP PBX. Seems to be a nice little unit. Support for only up to 50 users but a larger unit for 150 ppl is due out next year.

They are selling re-labeld Polycom phones and Counterpath eyeBeam (no video) for their softphone. They have an Adtran phone they are working on.

The system definately needs some software updates and should be up to speed by the November patch (early December).

The GUI is typical of the entire Adtran switch/router line. Looks pretty nice and easy to use.

See more about it here:

SIP Tapi

Had a customer the other day looking to do pops from Outlook. I found a couple cool products to do this.

Global IP has modified xten Pro a little bit to create a soft phone that has a Tapi driver. They call it X_TAPI Pro. Check it out here:

It seems to be able to pop contacts from the user's contacts folder only.

A more interesting product that can work with any Tapi driver is Identapop. Check it out here:

Identapop can pull contacts no only from your local contacts folder but also from public folders. It also has the ability to log calls to Outlooks Journal for historical call information.

SIP Tapi is an open source TAPI driver that seems to allow outbound dialing only. It can dial any contact in outlook.

Here's a wiki entry on how to setup SIP Tapi with sipxpbx:

Asterisk at Home Changes Name - Now trixbox

Asterisk at Home has changed its name!

The A@H project (formerly at is now called trixbox. The new web site is

For those not familliar, this project encompases CentOS Linux, Asterisk, FreePBX (formerly Asterisk Management Portal [AMP]) and a few other pieces of open source software.


Welcome to my blog.

I'll use this space to document my VoIP / IP Telephony research and rantings...